Re: [Live-devel] playSIP and stdout file not readable

2013-12-31 Thread Plischke, Markus
The file isnt empty: -rw-r--r-- 1 root root 487200 Dec 31 09:18 test.alaw I think you got me wrong, VLC CANT play the file. If i try to open it, it is 0 seconds long. Regards Markus Von: live-devel-boun...@ns.live555.com [mailto:live-devel-boun...@ns.live555.com] Im Auftrag von Ross Finlayson

Re: [Live-devel] playSIP and stdout file not readable

2013-12-31 Thread Ross Finlayson
Oops, I misread your last email. No, of course VLC can't play the stream, because VLC doesn't include a SIP client. So forget that... But the fact remains that the data that you're receiving *should* contain PCM a-law audio (1 channel, 8 bits-per-sample, 8000 Hz), because that's what your ser

Re: [Live-devel] playSIP and stdout file not readable

2013-12-31 Thread Ross Finlayson
First, I assume that your file is non-empty - i.e., you actually received data :-) Assuming that your file is non-empty, then it *should* be containing PCM a-law audio, because that's what the server (Asterisk) reported. The fact that VLC (which uses the LIVE555 RTSP/RTP client library) was ab

Re: [Live-devel] playSIP and stdout file not readable

2013-12-31 Thread Plischke, Markus
Hi Ross, thanks for you incredible fast answer: ffmpeg -i test.alaw -acodec pcm_alaw -ar 8000 -ac 1 bla.mp3 ffmpeg version 1.0.8 Copyright (c) 2000-2013 the FFmpeg developers built on Dec 30 2013 15:28:54 with gcc 4.7.3 (Gentoo 4.7.3-r1 p1.4, pie-0.5.5) configuration: --prefix=/usr --libdir=/

Re: [Live-devel] playSIP and stdout file not readable

2013-12-31 Thread Ross Finlayson
You appear to be receiving the audio correctly, so I suspect your problem is simply that you are not decoding it properly. The data in your file ("test.alaw") should be PCM a-law audio, 1 channel, with a sampling frequency of 8000 Hz. You probably need to tell "ffmpeg" explicitly what kind of

[Live-devel] playSIP and stdout file not readable

2013-12-31 Thread Plischke, Markus
Hi, ich have a little problem with playSIP. My goal ist o call a asterisk server in the same network and be member in a confbridge an record everything. So far, everything is working except reading the recording: Command: /usr/local/bin/playSIP -a -A 8 -u user secret sip:3@172.16.16.53:5060 > /